The present invention is related to the field of speech communication and particularly to means for providing improved detection and correction of errors due to noise, fading etc. during transmission without increasing the required bit rate.
In the field of digital communications, such as mobile and portable two-way communications, there are a number of possible channel conditions which adversely affect the quality of transmission. These include fading from shadowing, multipath reception, Doppler due to motion of a vehicle and other causes of signal strength variation. Under conditions of heavy channel utilization with low data bit rates, it becomes particularly important to combine essentially error-free decoding with the minimum required number of bits per unit of information.
When digital data is being transmitted, as from one computer to another, the most common way of dealing with erroneous signals is to resend the original signal. In other words, data signals are not in real time, therefore, if an error or errors are detected in a received signal, a return signal can be sent to the transmitting unit requesting a retransmission of the data. In the case of speech signals, which exist in real time, no such method can be utilized. It is necessary to include with the information necessary for decoding the speech information sufficient added information so that the receiving unit can detect any errors, and do the best possible job of correcting those errors, all while a user is perceiving the output speech signals.
Since there is little possibility of reducing the incidence of channel impairments, many methods of correcting errors have been developed in the field. In the Sub-Band Coding system with which the present invention would preferably operate, speech signals are encoded in frames, typically 20 to 30 msec in length, including two main kinds of information, the sample information and the side information necessary for proper decoding of the sample information. Certain of these bits have been found to be more susceptible to errors in transmission than others. It is these more vulnerable bits which have been the subject of various error control efforts with varying degrees of success. In the patent application cross-referenced above, the side information relative to each channel has been divided into the most-significant-bits (MSB) and the least-significant-bits (LSB) with the MSB protected and utilized in a manner which improves the "robustness" of the transmission.
Other systems have utilized various coding schemes in the effort to make the systems more robust. In a paper given by Ekemark, Ralth and Stjernvall at the Nordic Seminar on Digital Land Mobile Radiocommunications Feb. 5-7, 1985, at Espoo, Finland, "Modulation and Channel Coding in Digital Mobile Telephony", rather complicated calculations are shown for a high performance modulation system which is said to be well suited to operation under the conditions of Rayleigh fading and co-channel interference which exist in the mobile telephone environment. While little if anything is gained by the modulation system of this article in a stationary environment, the main disadvantage of the system is the high overhead requirement. There is still a need for a method of improving the accuracy of decoded speech by encoding which prevents most of the errors introduced during transmission from affecting the resultant quality of the decoded speech signals. It is not particularly difficult to do any of these things if there are sufficient bits available for "protection". The complicated task is to protect the signal to a high degree of accuracy with a minimum of added bits.